The invention relates to data transmission using existing telecommunications media, and in particular relates to a technique for encoding, transmitting, receiving, and decoding digital data using vocal sounds.
The demand today is for higher and higher rates of data communication. As networks continue to gain acceptance and favor, there is a continuing desire to transmit ever-increasing amounts of data across the transmission medium in a given amount of time. The main complaint against the current land-line telephone modem system is its low throughput. Today, higher digital connectivity data rates are being demanded by users for Internet access, telecommuting, video conferencing, and similar applications. This demand has led to many techniques and systems for increasing the data bandwidth, such as the use of integrated services digital networks (ISDN), and T-carrier services such as T1 and T3. ISDN transmits at a data rate of 128 kilobits/second (KBPS), while T1 transmits at a significantly higher 1.544 megabits/second (MBPS). One technology, referred to as digital subscriber line (DSL) technology, allows digital information to be transferred via existing copper-based twisted-pair telephone lines at rates as high as 6 MBPS. In other words, the increased demand for network solutions has propelled the need to maximize the data bandwidth.
In the cellular arena, current telecommunication systems include both analog and digital systems. Analog cellular systems, which currently dominate cellular transmission systems, suffer a variety of problems, including low bandwidth. As with the present telephone modem system, the main drawback of the current cellular modem system is its low throughput. Achieving the maximum throughput in the range of 9000 BPS is difficult, due to the noisy environment of the cellular modem system. Throughput is typically in the range of 4800 to 9600 bits per second (BPS). However, such emerging and existing data communication applications as effective wireless access to the Internet, the World Wide Web, and other information systems require the availability of large bandwidths to exchange information. Inventions such as that disclosed in U.S. Pat. No. 5,946,633, the complete disclosure of which is incorporated herein by reference, attempt to increase the bandwidth of analog cellular systems to support emerging and existing data communication applications requiring such large bandwidths.
While some strive to increase the bandwidths of existing telecommunications systems, others use the existing restricted bandwidth available on present land-line, cellular, and satellite telephone systems to transmit ordinary voice information.
Present telephone systems are designed to use low data-rate voice compression techniques for efficiently transmitting voice information on the existing restricted bandwidth. Human speech is typically digitized and compressed to enable the voice signal to be transmitted over a limited bandwidth channel over a relatively low bandwidth communications link, such as the public telephone system. The amount of compression, referred to as the xe2x80x9ccompression ratio,xe2x80x9d is inversely related to the bit rate of the digitized signal. More highly compressed digitized voice signals with relatively low bit rates, such as 2400 BPS, can be transmitted over relatively lower quality communications links with fewer errors than less compressed voice signals with higher bit rates, such as 4800 BPS or more.
Several methods are known for digitizing and compressing human speech. One example, linear predictive coding using ten reflection coefficients of the analog voice signal, known as LPC-10, produces compressed digitized voice at 2400 BPS in real time, i.e., with a fixed, bounded delay with respect to the analog voice signal. LPC-10e is a single-stage voice compression algorithm defined in federal standard FED-STD-1015, entitled xe2x80x9cTelecommunications: Analog to Digital Conversion of Voice by 2,400 Bit/Second Linear Predictive Coding,xe2x80x9d which is incorporated in its entirety herein by reference. Although LPC-10 is a xe2x80x9clossyxe2x80x9d compression procedure in that some information contained in the analog voice signal is discarded during compression, the amount of loss is generally slight, and the reconstructed voice signal is an intelligible reproduction of the original analog voice signal.
Various attempts have been made to increase the compression of the analog voice signal. For example, U.S. Pat. No. 5,742,930, the complete disclosure of which is incorporated herein by reference, discloses a multi-stage voice compression algorithm to increase the overall compression ratio between the incoming analog voice signal and the resulting digitized voice signal over that obtained using only a single compression stage, without sacrificing the intelligibility of the subsequently reconstructed analog voice signal. The greater compression allows speech to be transmitted over a channel having a much smaller bandwidth than would otherwise be possible, thereby allowing the compressed signal to be sent over lower quality communications links.
U.S. Pat. No. 5,546,395, the complete disclosure of which is incorporated herein by reference, discloses a method for communicating analog voice signals as digital data using standard telephone lines by compressing the digital data and placing the compressed data into packets for transfer over the telephone lines. A voice control digital signal processor (DSP) operates one of several speech compression algorithms which produce a scaleable amount of compression, the compression ratio being inversely proportional to the quality of the speech the compression algorithm is able to reproduce. The higher the compression, the lower the reproduction quality. The compression ratio is selected based on various factors, including the speed or data bandwidth on the available communications connection.
However, while inventors strive to increase the bandwidths of existing telecommunications systems and others attempt to increase the amount of voice data transmitted over the low speed, or data bandwidth, available on present telephone systems, low speed data transmission using the existing bandwidth is being overlooked. As disclosed in U.S. Pat. No. 5,559,799, the complete disclosure of which is incorporated herein by reference, modern modulator/demodulator apparatus, commonly known as xe2x80x9cmodems,xe2x80x9d are used widely for transmission of data in analog circuits which use a voice band. Prior art attempts to increase data using the existing bandwidth, and voice compression techniques fail to provide transmission of data in other bandwidths because voice compression does not provide for the transmission of traditional modem tones. Furthermore, as yet unforeseen emerging technologies will need a method of low bandwidth data transmission during their infancy, if not beyond.
Therefore, what is needed is a technique for encoding, transmitting, receiving, and decoding digital data using the low bandwidth voice modulation/encoding techniques currently available on existing telecommunications systems.
The present invention overcomes the limitations of the prior art by providing a technique for encoding, transmitting, receiving, and decoding digital data using the low bandwidth voice modulation/encoding techniques currently available on existing telecommunications systems.
The present invention provides a method and device for modulating digital data into traditional human vocal tract sounds upon which a voice compression system operates, and decompressing and demodulating the received signal. The invention also provides optional clocking data included in the modulated signal to provide self clocking for incorporation into telecommunications systems having variable delay from transmitter to receiver, such as one of the known satellite communications systems available for air transport applications. The invention further provides optional error correction data that is optionally encoded into the modulated signal for robustness. The modulation/demodulation method of the invention provides a dictionary of voiced and unvoiced vocal sounds distinguishable by existing voice encoder/decoders, commonly referred to as xe2x80x9cvocoders.xe2x80x9d Digital data are preferably modulated and demodulated according to this dictionary.